[..] |
ASTERISK-25000: Deadlock in ast_do_masquerade (specifically in ast_hangup on the zombie clone if it's hungup during the masquerade) |
ASTERISK-25001: SendFAX error |
ASTERISK-25002: [patch] - Demote "Extension Changed" messages in Asterisk CLI |
ASTERISK-25003: Asterisk crashes on attended transfer (using feature) |
ASTERISK-25004: Crash in authenticated reinvite after originated T.38 FAX |
ASTERISK-25005: MixMonitor doesn't record outgoing calls |
ASTERISK-25006: [patch] Add support set character for quoted identifiers |
ASTERISK-25007: Notify packet to private IP endpoint behind nat with pjsip tls transport |
ASTERISK-25008: While Redirecting dual channel and one of them is AGENT channel. Agent loggs off form the system |
ASTERISK-25009: DNS Tests: SRV Priority |
ASTERISK-25010: DNS Tests: Failover order |
ASTERISK-25011: DNS Tests: Failover to A/AAAA |
ASTERISK-25012: DNS Tests: NAPTR Nominal - Correct Order |
ASTERISK-25013: DNS Tests: NAPTR Nominal - Correct Preference |
ASTERISK-25014: DNS Tests: NAPTR Nominal - Restricted Transport |
ASTERISK-25015: DNS Tests: NAPTR Nominal - Failover of preferences |
ASTERISK-25016: one-sided audio when auto_comedia is set |
ASTERISK-25017: download from svn fails |
ASTERISK-25018: pjsip show endpoints crashes asterisk when qualified aors present |
ASTERISK-25019: called party Unhold SRTP calls cause NO MEDIA. unprotect failed with: authentication failure 110 |
ASTERISK-25020: Mismatched response to outgoing REGISTER request |
ASTERISK-25021: Fix invalid pointer dereference on module load |
ASTERISK-25022: Memory leak setting up DTLS/SRTP calls |
ASTERISK-25023: Deadlock in chan_sip in update_provisional_keepalive |
ASTERISK-25024: there is no ice when i make originale |
ASTERISK-25025: Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13. |
ASTERISK-25026: Git conversion: Non-C files not switched to ASTERISK_REGISTER_FILE |
ASTERISK-25027: Build System: Many ARI modules are missing dependencies. |
ASTERISK-25028: Build System: Unneeded defines in asterisk/buildopts.h |
ASTERISK-25029: Astobj2: Create ao2_weakproxy_ref_object function. |
ASTERISK-25030: Avoid using LIKE when using realtime engine |
ASTERISK-25031: DTMF INFO before answer leads to 200 OK without Contact: header |
ASTERISK-25032: [patch]cel_odbc sometimes inserts CEL with wrong eventtime |
ASTERISK-25033: Asterisk 13 (branch head) won't compile without PJSip |
ASTERISK-25034: chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests. |
ASTERISK-25035: Stuck Channels |
ASTERISK-25036: Asterisk still send an INVITE request after a call was canceled(realtime, rtcachefriends is enabled) |
ASTERISK-25037: res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message |
ASTERISK-25038: Queue log "EXITWITHTIMEOUT" does not always contain waiting time |
ASTERISK-25039: getting major delays when connecting a call to a webrtc client |
ASTERISK-25040: pbx: Improve performance of reloads by making hint destruction more performant |
ASTERISK-25041: [patch]Broken column type checking in res_config_mysql addon |
ASTERISK-25042: asterisk.conf options override command-line options. |
ASTERISK-25043: [patch] Avoiding ERR_remove_state in OpenSSL |
ASTERISK-25044: sorcery: Add ability to insert a new wizard into an object type's list |
ASTERISK-25045: vector: Add new capabilities and unit tests |
ASTERISK-25046: Chan_sip deadlock persistent between Asterisk 1.8 versions, using Realtime. |
ASTERISK-25047: Call remaining in "core show channels" after hangup |
ASTERISK-25048: Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled. |
ASTERISK-25049: CLI: Enable automatic references to modules |
ASTERISK-25050: res_pjsip cannot load when contact is enabled for realtime |
ASTERISK-25051: Remove unneeded uses of optional_api providers. |
ASTERISK-25052: OPTIONAL_API: Remove ABI option. |
ASTERISK-25053: Unit test category /main/presence missing trailing slash. |
ASTERISK-25054: Formats interface's cannot be unregistered, needs to hold modules until shutdown. |
ASTERISK-25055: ARI Snoop Channel: Improve documentation |
ASTERISK-25056: Modules: Make ast_module_info->self available to auxiliary sources. |
ASTERISK-25057: res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree |
ASTERISK-25058: res_pjsip_pubsub: Crash in ast_sorcery_hash called from subscription_persistence_update |
ASTERISK-25059: recording calls in ogg format creates memory leak |
ASTERISK-25060: configure - libxml2 not found |
ASTERISK-25061: pbx_config: Register manager actions with module version of macro. |
ASTERISK-25063: [patch]add X.509 subject alternative name support to Asterisk TLS support |
ASTERISK-25064: Members (ringinuse disabled) of multiple queues ringing with other queue calls. |
ASTERISK-25065: SRTP failing over time |
ASTERISK-25066: DTMFs are not sent to the bridge channel if they are used by any built-in or dynamic feature |
ASTERISK-25067: Sorcery Caching: Implement a new caching module |
ASTERISK-25068: Move commonly used FreePBX extra sounds to the core set |
ASTERISK-25069: main/message.c: Fix unregister functions |
ASTERISK-25070: Fix FTBFS on Hurd |
ASTERISK-25071: RFC3581 compliance |
ASTERISK-25072: res_pjsip_outbound_registration: line functionality. Additional check for using the request URI |
ASTERISK-25073: Asterisk crash by ast_format_cap_append |
ASTERISK-25074: Regression: Recent clang-related change broke cross compiling of Asterisk |
ASTERISK-25075: func_speex.c does not compile, missing speex/speex_preprocess.h |
ASTERISK-25076: res_pjsip: Failover does not occur on connection-less transport or 503 response |
ASTERISK-25077: use IP_FREEBIND on network sockets |
ASTERISK-25078: sig_pri: Publish progress codes. |
ASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel |
ASTERISK-25080: res_pjsip_refer: Refer code invoked with unexpected NULL channel on session |
ASTERISK-25081: res_pjsip_refer: Refer code invoked with unexpected NULL channel on session |
ASTERISK-25082: Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox. |
ASTERISK-25083: Message.c: Message channel becomes saturated with frames leading to spammy log messages |
ASTERISK-25084: silenceSupp is not sent by default |
ASTERISK-25085: [patch]Potential crash after unload of func_periodic_hook or test_message |
ASTERISK-25086: [patch]PJSIP crashes if endpoint missing in Dial() |
ASTERISK-25087: Asterisk segfault when using Directory application with alias option and specific mailbox configuration |
ASTERISK-25088: res_pjsip: Failure to specify cipher in a TLS transport causes a SIGABRT in pjproject |
ASTERISK-25089: res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly |
ASTERISK-25090: CLI core show channel truncates cdr variables |
ASTERISK-25091: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge |
ASTERISK-25092: xmpp connection failure causes continuous error messages |
ASTERISK-25093: Asterisk stop working suddenly often |
ASTERISK-25094: PBX core: Investigate thread safety issues |
ASTERISK-25095: Not manage "busy/congestion" in Asterisk release after v. 1.8.29.0 |
ASTERISK-25096: [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h) |
ASTERISK-25097: Asterisk13.3.2 PJSIP configuration user_agent doesn't go into effect |
ASTERISK-25098: Auto-close for JIRA issues with merged commits is not functioning |
ASTERISK-25099: res_rtp_asterisk: Crash when using DTLS |
ASTERISK-25100: asterisk coredump if host has an IPv6 address that end with ::80 |
ASTERISK-25101: DTLS configuration can not be specified in the general section - documentation |
ASTERISK-25102: res_config_odbc / libodbc - Asterisk core dumps with signal 6 when connection lost during query |
ASTERISK-25103: Roundup - investigate Asterisk DTLS crashes |
ASTERISK-25104: Unnecessary Unlink event on reINVITE when using Monitor() |
ASTERISK-25105: res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4 |
ASTERISK-25106: Segmentation fault in ast_variable_new when using app_voicemail with realtime |
ASTERISK-25107: Phone losing registration 10 seconds after reboot. |
ASTERISK-25108: configure check for older unbound library |
ASTERISK-25109: [patch] CEL and CDR - Assigned separator for column name and values. |
ASTERISK-25110: res_resolver_unbound.c compilation failure: SIGURG is undeclared in func unbound_resolver_stop |
ASTERISK-25111: Playback fails when matching h264 codec |
ASTERISK-25112: Logger: Configuration settings are not reset to default during reload. |
ASTERISK-25113: install_prereq in Debian 8 without "standard system utilities" |
ASTERISK-25114: res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes |
ASTERISK-25115: Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c |
ASTERISK-25116: res_pjsip: Two PeerStatus AMI messages are sent for every status change |
ASTERISK-25117: res_mwi_external_ami: Fix manager action registrations. |
ASTERISK-25118: [patch]PeerStatus Event Unsubscribed with Peer IP Address |
ASTERISK-25119: Crash on pjsip_tls_transport_start2 |
ASTERISK-25120: Astobj2: Weakproxy subscriptions should be run in reverse order. |
ASTERISK-25121: Stasis: Fix unsafe use of stasis_unsubscribe in modules. |
ASTERISK-25122: Large SIP packet received via pjsip over websocket crashes Asterisk |
ASTERISK-25123: Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP |
ASTERISK-25124: menuselect configure unnecessarily requires libxml2 for 11.18.0-rc1 |
ASTERISK-25125: Testsuite: Investigate lock-up of tests/fastagi/record-file |
ASTERISK-25126: process_sdp: Can't provide secure audio requested in SDP offer |
ASTERISK-25127: DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending |
ASTERISK-25128: Datastore: Implement automatic module references. |
ASTERISK-25129: wrong automatic ras address assignment if multihomed |
ASTERISK-25130: Load average very high with Local channels |
ASTERISK-25131: chan_pjsip: In-dialog authentication not handled. |
ASTERISK-25132: escaping manually |
ASTERISK-25133: PCI Voice/Data/Fax Modem |
ASTERISK-25134: Problem with CDR information with incoming calls |
ASTERISK-25135: [patch]RTP Timeout hangup cause code missing |
ASTERISK-25136: gosub issue |
ASTERISK-25137: endpoint stasis messages are delivered twice |
ASTERISK-25138: Unclosed parenthesis in AGI argument leads to further arguments concatenated - parameter quoting not respected |
ASTERISK-25139: Malicious transfer sequence locks up Asterisk |
ASTERISK-25141: pjsip_options: Contact reference leak |
ASTERISK-25142: G722 recordings and spy does not give audio in asterisk => 11.17 |
ASTERISK-25143: segmentation fault in ast_format_get_type |
ASTERISK-25144: Testsuite: Investigate failures in tests for channels/pjsip/subscriptions/rls |
ASTERISK-25145: Unable to record calls picked up from parking lot |
ASTERISK-25146: DNS: Create system level resolver |
ASTERISK-25147: [patch]Testsuite: tests/pjsip/transfers/blind_transfer/caller_with_hold fails after PJSIP 2.4 upgrade |
ASTERISK-25148: res_pjsip NULL channel audit |
ASTERISK-25149: res_pjsip_session: Attended transfer nominal callee local direct media test failing |
ASTERISK-25150: chan_pjsip: RLS subscriptions produce leaks |
ASTERISK-25151: results of the automated codereview of the Asterisk project |
ASTERISK-25152: func_cdr: Deadlock when used with ARI originated channels |
ASTERISK-25153: Asterisk originate with Application/data does not go to dialplan |
ASTERISK-25154: [patch]fromtag may need to be updated after successful call dialog match |
ASTERISK-25155: Musiconhold bug second time caller is put on hold |
ASTERISK-25156: chan_pjsip’s CHAN_START cel event lacks the correct context and exten |
ASTERISK-25157: bridging: Performing a blonde transfer does not result in connected line updates |
ASTERISK-25158: res_pjsip: Add option to use AAL2 packing when negotiating g.726 |
ASTERISK-25159: queue_log does not log PAUSE events for pause states set with QUEUE_MEMBER |
ASTERISK-25160: [patch] Opus Codec: SIP/SDP line fmtp missing when called internally |
ASTERISK-25161: PJSIP "connected_line_method=update" segfault in libpjmedia.so during attended transfer |
ASTERISK-25162: func_pjsip_aor: Leak of contact in iterator |
ASTERISK-25163: Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback |
ASTERISK-25164: asterisk 11.9 |
ASTERISK-25165: Testsuite - Sorcery memory cache leaks |
ASTERISK-25166: No audio when using direct media and a codec with a dynamic payload |
ASTERISK-25167: Testsuite: Resolve remaining Asterisk shutdown timeout's |
ASTERISK-25168: Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c |
ASTERISK-25169: No audio from voicemail app with v13.4.0 on Grandstream GXP20XX phones |
ASTERISK-25170: Segfault in call to vsnprintf from astman_append |
ASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound. |
ASTERISK-25172: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request |
ASTERISK-25173: ARI: Add the ability to load/reload/unload an Asterisk module |
ASTERISK-25174: Wiki Documentation - Features - Recording (one touch) features |
ASTERISK-25175: Wiki Documentation - Features - Application Map & Dynamic Features |
ASTERISK-25176: Asterisk segfault on reload res_odbc.so |
ASTERISK-25177: Make configure script bail if pjproject was built statically |
ASTERISK-25178: Cannot login to Asterisk Forums |
ASTERISK-25179: CDR(billsec,f) and CDR(duration,f) report incorrect values |
ASTERISK-25180: res_pjsip_mwi: Unsolicited MWI requires reload |
ASTERISK-25181: ARI: Channels added to Stasis application during WebSocket creation don't receive a StasisStart event |
ASTERISK-25182: [patch] on CLI sip reload, new codecs get appended only |
ASTERISK-25183: PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel |
ASTERISK-25184: Lockup in IAX channel |
ASTERISK-25185: Segfault in app_queue on transfer scenarios |
ASTERISK-25186: Testing |
ASTERISK-25187: Caller or member information missing from app_queue AgentComplete event |
ASTERISK-25188: Fax Detection through RFC2833 |
ASTERISK-25189: AMI: Add Linkedid header to standard channel snapshot information. |
ASTERISK-25190: Testing 2 |
ASTERISK-25191: Testing 3 |
ASTERISK-25192: Testing 4 |
ASTERISK-25193: Testing 5 |
ASTERISK-25194: Incorrect GotoIf Behavoir |
ASTERISK-25195: Can I? |
ASTERISK-25196: res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present |
ASTERISK-25197: asterisk 13.4.0 fails to transcode calls to/from g729 with TCE400P |
ASTERISK-25198: chan_dahdi with SS7 is writing log messages to console despite verbose 0 |
ASTERISK-25199: Testing Again |
ASTERISK-25200: Wanna see something |
ASTERISK-25201: Crash in PJSIP distributor on already free'd threadpool |
ASTERISK-25202: Hints extension state broken between 13.3.2 and 13.4 |
ASTERISK-25203: AMI event QueueCallerLeave not always sent |
ASTERISK-25204: res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs. |
ASTERISK-25205: pjsip Via broken if there are multiple transports with different bind ports |
ASTERISK-25206: No ability to control asterisk if permission failure on /var/run |
ASTERISK-25207: Yolo |
ASTERISK-25208: Testing |
ASTERISK-25209: asterisk.org still refers to SVN - update references to SVN in documentation and website |
ASTERISK-25210: pjsip 'qualify_timeout' problem |
ASTERISK-25211: Prevent timer_pthread from loading with non-compliant systems |
ASTERISK-25212: [patch]Segfault when using DEBUG_FD_LEAKS |
ASTERISK-25213: [patch]Possibility of deadlock in chan_sip INVITE early Replace code |
ASTERISK-25214: DTMF over SIP INFO and direct media does not work well together |
ASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember |
ASTERISK-25216: Asterisk periodic hangs. UDP Recv-Q greatly exceeds zero. |
ASTERISK-25217: [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c |
ASTERISK-25218: res_config_pgsql freezes asterisk at startup - "init table" request (function find_table) running infinitely |
ASTERISK-25219: [patch]Source and destination overlap in memcpy in rtp_engine.c |
ASTERISK-25220: [patch]Closing of fd -1 in chan_mgcp.c |
ASTERISK-25221: [patch]Improvement to "sip show peer <peer>" - Get additional contact parameters |
ASTERISK-25222: Crash in recurring cancel callback called from ast_dns_resolve_cancel on junk pointer |
ASTERISK-25223: Crash in RLS MWI termination tests |
ASTERISK-25224: WARNING message flooding Asterisk logs |
ASTERISK-25225: testsuite: lua tests using SIPp cannot run on Jenkins |
ASTERISK-25226: chan_sip: Channel leak in branch 13 on early replaces call pickup |
ASTERISK-25227: No audio at in-band announcements in ooh323 channel |
ASTERISK-25228: configure with netsnmp fail |
ASTERISK-25229: Exchanging Device and Mailbox State Using PJSIP fails after restart of peer |
ASTERISK-25230: Crash in channels/pjsip/basic_calls/incoming/off-nominal/userpass when decreasing reference on PJSIP transport |
ASTERISK-25231: Set Music AGI command with class argument fails silently - need user level log message |
ASTERISK-25232: unistim not showing callerid |
ASTERISK-25233: Wiki Documentation - Configuration/Reporting - Call Detail Records |
ASTERISK-25234: Wiki Documentation - Configuration/Reporting - Channel Event Logging |
ASTERISK-25235: Wiki Documentation - Configuration/Interfaces - Calendaring |
ASTERISK-25236: Asterisk reject incoming call from FXO gateway |
ASTERISK-25237: stasis_cache.c:845 caching_topic_exec: - misleading ERROR message |
ASTERISK-25238: ARI: Support push configuration |
ASTERISK-25239: Asterisk 13.4.0 crashes |
ASTERISK-25240: bridge_native_rtp: Direct media wrongfully started when completing attended transfer |
ASTERISK-25241: Wiki Documentation - Configuration/Interfaces - Asterisk Gateway Interface |
ASTERISK-25242: PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'? |
ASTERISK-25243: Asterisk logger can't log level "security" to syslog |
ASTERISK-25244: WIki Documentation - Configuration/Interfaces - Asterisk Manager Interface |
ASTERISK-25245: Wiki Documentation - Organize and update database connectivity and realtime documentation |
ASTERISK-25246: Queues repeatedly try agents in Unavailable status |
ASTERISK-25247: choppy audio when spying on a g722 channel, chan_sip or chan_pjsip |
ASTERISK-25248: [patch]Improve Chan_Local's bridging speed |
ASTERISK-25249: Features code not working for called party when Local channels are involved |
ASTERISK-25250: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback |
ASTERISK-25251: getifaddrs() blocks infinitely in PJSIP |
ASTERISK-25252: ARI: Add the ability to manipulate log channels |
ASTERISK-25253: confbridge volume options and other volume controls such as func_volume don't work |
ASTERISK-25254: Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. |
ASTERISK-25255: Missing AMI VarSet events when setting to an empty string. |
ASTERISK-25256: [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable. |
ASTERISK-25257: [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope |
ASTERISK-25258: chan_pjsip: Incorrect format switch on received RTP packet |
ASTERISK-25259: chan_pjsip: Add rtptimeout support |
ASTERISK-25260: Wiki Documentation - Parking! |
ASTERISK-25261: Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User' |
ASTERISK-25262: Memory leak when a caller channel does multiple dials and CEL is enabled |
ASTERISK-25263: [patch]cdr_adaptive_odbc: CDR insert failure due to reversed if logic |
ASTERISK-25264: Crash in ast_chan_setoption |
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 |
ASTERISK-25266: Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate |
ASTERISK-25267: [patch] Alembic - broken compatibility for Oracle and Microsoft SQL |
ASTERISK-25268: Neither a src change or marker after (attended) transfer |
ASTERISK-25269: Speex VAD: SIGSEGV when softmixing stasis-bridges |
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind |
ASTERISK-25271: Parking & blind transfer: Transferer channel not hung up if no MOH |
ASTERISK-25272: [patch]The ICONV dialplan function sometimes returns garbage |
ASTERISK-25273: Can't compile Asterisk - 'make menuselect' fails with segfault and core dump |
ASTERISK-25274: A11 SIGSEGV 'Double free or corruption' in backtrace from pj_pool_release (sip_destroy -> pj_ice_sess_destroy) |
ASTERISK-25275: A11 SIGSEGV from pjnpath check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt) |
ASTERISK-25276: "confbridge record" makes # of asterisk subprocess growing |
ASTERISK-25277: Asterisk forwards DTMF when call is on hold |
ASTERISK-25278: pjsip: MWI aggregate test is failing consistently. |
ASTERISK-25279: Deadlock using chan_sip |
ASTERISK-25280: Reset service Asterisk: ERROR astobj2.c: user_data is NULL |
ASTERISK-25281: PJSIP, ODBC and Oracle - case sensitive field name checks in sorcery break Oracle compatibility |
ASTERISK-25282: rtptimeout does not kick in because of zero sized frames (lost packets) |
ASTERISK-25283: res_ari: Multiple PJSIP contacts can't be dialed directly |
ASTERISK-25284: Parking & blind transfer: Transferer channel not hung up if no MOH |
ASTERISK-25285: app_voicemail prompts play improperly for Portuguese language |
ASTERISK-25286: After a period of time chan_iax2 stops accepting packets |
ASTERISK-25287: Busy Detect not working |
ASTERISK-25288: Configure fails to detect uuid-devel on Fedora 22 |
ASTERISK-25289: Build System does not respect CFLAGS and CXXFLAGS when building menuselect |
ASTERISK-25290: Build System does not respect CFLAGS and CXXFLAGS placed on the command line |
ASTERISK-25291: Test suite not available via `make check` or `make test` |
ASTERISK-25292: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails |
ASTERISK-25293: Crash in DNS core if no DNS result set on query |
ASTERISK-25294: srtp's crypto_get_random deprecated |
ASTERISK-25295: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h |
ASTERISK-25296: RTP performance issue with several channel drivers. |
ASTERISK-25297: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests |
ASTERISK-25298: pjsip: subscriptions/mwi/unsolicited/mailbox_count_changes sporadically failing |
ASTERISK-25299: RTP port leaks with incoming OOH323 calls |
ASTERISK-25300: ConfBridge() in 13.1-cert2 is not combining configuration from static text config file with dynamic config in dialplan with CONFBRIDGE function. |
ASTERISK-25301: asterisk segfault in res_hep_pjsip.so on client connect |
ASTERISK-25302: Error:Oh dear... we couldn't allocate a port for RTP instance |
ASTERISK-25303: Error 4 in app_queue.so in 13.1-cert2 |
ASTERISK-25304: res_pjsip: XML sanitization may write past buffer |
ASTERISK-25305: Dynamic logger channels can be added multiple times |
ASTERISK-25306: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes. |
ASTERISK-25307: Hangup on channel using FastAGI does not hang up child channels |
ASTERISK-25308: ari: Websocket leak |
ASTERISK-25309: [patch] iLBC 20 advertised |
ASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED |
ASTERISK-25311: Asterisk may restart if using cdr_odbc + MySQL in strict mode |
ASTERISK-25312: res_http_websocket: Terminate connection on fatal cases |
ASTERISK-25313: tests/bridge/connected_line_update: Sporadically failing |
ASTERISK-25314: tests/rest_api/asterisk/logging/get_logging: Consistently failing |
ASTERISK-25315: DAHDI channels send shortened duration DTMF tones. |
ASTERISK-25316: PJSIP qualify in mutlihomed system sent from wrong transport |
ASTERISK-25317: asterisk sends too many stun requests |
ASTERISK-25318: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing |
ASTERISK-25319: tests/rest_api/asterisk/logging/rotate_log / add_log: Sporadically failing |
ASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite |
ASTERISK-25321: [patch]DeadLock ChanSpy with call over Local channel |
ASTERISK-25322: Crash occurs when using MixMonitor with t() or r() options. |
ASTERISK-25323: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG |
ASTERISK-25324: MAKE CALL /Recieve Call from Web client in asterisk 1.8 |
ASTERISK-25325: ARI PUT reload chan_sip HTTP response 404 |
ASTERISK-25326: asterisk-ari: ARI API to stop running music |
ASTERISK-25327: Annoying ERROR: lock.c:459 __ast_pthread_mutex_unlock: app_queue.c line 6445 (try_calling): mutex 'qe->chan' freed more times than we've locked! |
ASTERISK-25328: Wrong ANSWEREDTIME after Dial app execution |
ASTERISK-25329: Asterisk configure fails on 'cannot find ptlib-config', despite ptlib-config existing |
ASTERISK-25330: Failed to originate a call using AMI on OOH323 channel (Avaya) |
ASTERISK-25331: install_prereq is not installing sqlite 3 library on CentOS |
ASTERISK-25332: marker bit lost in outgoing stream when incoming stream has vad |
ASTERISK-25333: [patch]App meetme sometimes not close fd with pseudo module |
ASTERISK-25334: Testing |
ASTERISK-25335: Testing bob |
ASTERISK-25336: Documentation: Explanation of DIALEDTIME and ANSWEREDTIME variables could be more explicit |
ASTERISK-25337: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub |
ASTERISK-25338: Failed to authenticate device messages don't report connection ip |
ASTERISK-25339: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid |
ASTERISK-25340: Manager.conf TLS doesn't activates |
ASTERISK-25341: bridge: Hangups may get lost when executing actions |
ASTERISK-25342: res_pjsip: Repeated usage of pj_gethostip may block |
ASTERISK-25343: Successful attended transfer on queue leaves agent with Music On Hold |
ASTERISK-25344: Random crashes (SIGSEGV / SIGABRT) on Asterisk 13.1.0~dfsg-1+b1 (possibly in combination with stasis) |
ASTERISK-25345: pjsip:0 <?>: tsx0xb3fe5d1c ...Failed to send Request msg INVITE/cseq=28532 (tdta0xb6bb49c0)! err=171060 (Unsupported transport (PJSIP_EUNSUPTRANSPORT)) |
ASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on call pickup |
ASTERISK-25347: Crash on res_odbc reload in SOCK_flush_output from /usr/lib64/psqlodbc.so |
ASTERISK-25348: Asterisk Crashes after caller records name with the Privacy Options |
ASTERISK-25349: Asterisk freeze due to schedule ID cancellation failure |
ASTERISK-25350: Asterisk crashes |
ASTERISK-25351: SIP INFO not ACK'd by chan_pjsip on Asterisk 13.5.0 |
ASTERISK-25352: res_hep_rtcp correlation_id is different then res_hep |
ASTERISK-25353: [patch] Transcoding while different in Frame size = Frames lost |
ASTERISK-25354: Unable to compile any version |
ASTERISK-25355: sched: ast_sched_del may return prematurely due to spurious wakeup |
ASTERISK-25356: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist |
ASTERISK-25357: AMI GetConfigJSON |
ASTERISK-25358: dateformat not read from logger.conf by remote console |
ASTERISK-25359: Leak when not Answering |
ASTERISK-25360: chan_sip:wrong ice candidate fails html5client on mobile connection |
ASTERISK-25361: [patch]possible null pointer issue in chan_iax2 - with potential fix |
ASTERISK-25362: Deadlock due to presence state callback |
ASTERISK-25363: PJSIP/rls and Testsuite: channels/pjsip/subscriptions/rls/lists/off_nominal/large_notify reports success after reactor times out and terminates the unfinished sipp scenario |
ASTERISK-25364: [patch]Issue a TCP connection(kernel) and thread of asterisk is not released |
ASTERISK-25365: Persistent subscriptions have extra Content-Length/corrupted messages |
ASTERISK-25366: Segmentation fault - in ast_manager_build_channel_state_string_prefix at manager_channels.c:417 |
ASTERISK-25367: pbx: Long pattern match hints may cause "core show hints" to crash |
ASTERISK-25368: English sound prompt no-valid-responce-transfering has an incorrect name and is mis-matched vs other language sets |
ASTERISK-25369: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel |
ASTERISK-25370: res_corosync segfaults at startup with corosync version > 2.x |
ASTERISK-25371: Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event |
ASTERISK-25372: SIP/2.0 401 Unauthorized for incoming calls. |
ASTERISK-25373: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants |
ASTERISK-25374: Crash in CDR handle_dial_message where peer is null |
ASTERISK-25375: Bad ao2 pointer on snapshot cleanup after creation |
ASTERISK-25376: Scripts: check file versions for Asterisk and dependencies |
ASTERISK-25377: res_pjsip: Change default "From user" from UUID to something more palatable |
ASTERISK-25378: Segfault in pj_pool_alloc () from /usr/lib/libpj.so.2 |
ASTERISK-25379: no sound on pjsip channel with bridge_native_rtp enabled |
ASTERISK-25380: Oh dear... we couldn't allocate a port for RTP instance |
ASTERISK-25381: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts |
ASTERISK-25382: segfault in ast_json_free (p=0x77f9d88555b8) at json.c:190 |
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG enabled |
ASTERISK-25384: Regular Asterisk crashes when using Page application. "user_data is NULL" |
ASTERISK-25385: A11 SIGSEGV Crashes in srtp_get_stream (), in ast_srtp_protect |
ASTERISK-25386: Asterisk Chan_sip.c Deadlock. All SIP traffic stops |
ASTERISK-25387: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten |
ASTERISK-25388: chan_sip partial unresponsive, one permanent lock |
ASTERISK-25389: pjsip: crash on null uri in contact header |
ASTERISK-25390: default_from_user can crash with certain configuration backends |
ASTERISK-25391: AMI GetConfigJSON returns invalid JSON |
ASTERISK-25392: A11 Deadlock detected, full thread bt |
ASTERISK-25393: Non-realtime and Realtime can not exist together |
ASTERISK-25394: pbx: Incorrect device and presence state when changing hint details |
ASTERISK-25395: Crash when establishing subscription with pjsip |
ASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP |
ASTERISK-25397: [patch]chan_sip: File descriptor leak with non-default timert1 |
ASTERISK-25398: chan_iax2 call failed to authenticate |
ASTERISK-25399: app_queue: AgentComplete event has wrong reason |
ASTERISK-25400: Hints broken when "CustomPresence" doesn't exist in AstDB |
ASTERISK-25401: Segmentation-fault crash within pjnath |
ASTERISK-25403: A11 SIGSEGV clearerr (fp=0x0) at clearerr.c:26...in ast_careful_fwrite |
ASTERISK-25404: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c |
ASTERISK-25405: [patch] CLI: core show fd: add timestamp |
ASTERISK-25406: Misc anomalies in Swagger definitions |
ASTERISK-25407: Asterisk fails to log to multiple syslog destinations |
ASTERISK-25408: One RTP stream is lost out of the NIC for approx 5 sec then returns |
ASTERISK-25409: Asterisk not reading entire TLSCERTFILE |
ASTERISK-25410: app_record: RECORDED_FILE variable not being populated |
ASTERISK-25411: PJSIP functionality becomes unresponsive after some time |
ASTERISK-25412: Blanks in database fields leads to unexpected results |
ASTERISK-25413: res_pjsip: does not have IP only endpoint identification per-endpoint - indentify_by work-around |
ASTERISK-25414: CLONE - [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized |
ASTERISK-25415: A11 SIGSEGV 'Double free or corruption' in backtrace from pj_pool_release |
ASTERISK-25416: pbx_dundi: Using scheduler context after deletion |
ASTERISK-25417: res_fax - Asterisk does not send localstationidentifier to remote site |
ASTERISK-25418: On-hold channels redirected out of a bridge appear to still be on hold |
ASTERISK-25419: Dialplan Application for Integration of StatsD |
ASTERISK-25420: Some documentation issues for AMI |
ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending |
ASTERISK-25422: [patch]The 'Bridge' event always reports the 'Bridgetype' as 'core' even if it's 'native' |
ASTERISK-25423: Caller gets no Connected line update during call pickup. |
ASTERISK-25424: asterisk.conf syntax error causes inscrutable crash |
ASTERISK-25425: logger: Add JSON structured logging |
ASTERISK-25426: Core dump in CDR handler |
ASTERISK-25427: Callerid change does not always emit NewCallerid AMI event |
ASTERISK-25428: Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX |
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames |
ASTERISK-25430: Testsuite: batched rls subscription failure |
ASTERISK-25431: A11 SIGSEGV in check_cached_response (ast_rtcp_read -> pj_stun_session_on_rx_pkt) |
ASTERISK-25432: Testsuite: tests/channels/SIP/tcpauthlimit/tcp_client_scenario is Failing |
ASTERISK-25433: Crash on CLI command 'data get' |
ASTERISK-25434: Compiler flags not reported in 'core show settings' despite usage during compilation |
ASTERISK-25435: Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero. |
ASTERISK-25436: Segmentation fault relating to JSON, stasis, and fax |
ASTERISK-25437: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c |
ASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not enabled |
ASTERISK-25439: Segfault in find_entry () from /usr/lib/libpj.so.2 (dns_resolver, qualify_contact) |
ASTERISK-25440: Asterisk does not use owner parameter in SDP breaking RFC3264 |
ASTERISK-25441: Deadlock in res_sorcery_memory_cache. |
ASTERISK-25442: using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c) |
ASTERISK-25443: [patch]IPv6 - Potential issue in via header parsing |
ASTERISK-25444: [patch]Music On Hold Warning misleading |
ASTERISK-25445: AMI QueueStatus broken |
ASTERISK-25446: Warnings with jitter buffer enabled and transcoding from G722 to ulaw |
ASTERISK-25447: Wrong template file 'realtime.sql' in contrib/realtime/postgresql |
ASTERISK-25448: Motor Sound after pickup incoming call |
ASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny |
ASTERISK-25450: Multiple realtime entries for same (ldap)driver in sorcery.conf not working anymore. |
ASTERISK-25451: Broken video - erased rtp marker bit |
ASTERISK-25452: Registering with Caret character at the end of password |
ASTERISK-25453: user_agent not set in Server header |
ASTERISK-25454: Crashing possibly related to cdr_manager |
ASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql |
ASTERISK-25456: Asterisk becomes unresponsive - res_rtp_asterisk: WebRTC STUN Deadlock |
ASTERISK-25457: Chan_PJSIP No MoH / Hold |
ASTERISK-25458: Unable to set CDR variable in h extension or hangup_handler |
ASTERISK-25459: New dependency on libxml2 |
ASTERISK-25460: UDP leak |
ASTERISK-25461: Nested dialplan #includes don't work as expected. |
ASTERISK-25462: pjsip show channels segfault: Address 0x2 out of bounds in res_sorcery_realtime.c |
ASTERISK-25463: Starting Asterisk with ARI module started but sorcery not causes pain |
ASTERISK-25464: Segfault with T.38 protocol and ReceiveFax Application |
ASTERISK-25465: script for migrating realtime chan_sip tables to pjsip |
ASTERISK-25466: pjsip: Endpoints added to pjsip.conf during runtime - reload results in an 'invalid' state for all but the last endpoint loaded |
ASTERISK-25467: chan_sip/webrtc Asterisk + Chrome M47 consistent 0.9s ice handshake delay since commit 1ad827 |
ASTERISK-25468: Deadlock in chan_sip - core show locks shows do_monitor lock |
ASTERISK-25469: chan_sip enabled unnegotiated session-timers after reINVITE if session-minse is non-default. |
ASTERISK-25470: SDP version increased when no change in SDP on 183 session progress retransmit |
ASTERISK-25471: [patch]Add subscribe_context to res_pjsip |
ASTERISK-25472: Swagger scripts are not replacing format variable in file brief |
ASTERISK-25473: Test infrastructure: Need to reimplement coverage reports |
ASTERISK-25474: Test infrastructure: Need to reimplement REF_DEBUG testing |
ASTERISK-25475: PJSIP tab completion of 'pjsip show endpoint' results in query storm and takes a very long time |
ASTERISK-25476: chan_sip loses registrations after a while |
ASTERISK-25477: pjsip show "command" like [criteria] |
ASTERISK-25478: Need configuration option(s) to control how res_pjsip_endpoint_identifier_user performs endpoint lookup |
ASTERISK-25479: Allow CDR's to be modified before being dispatched to engines |
ASTERISK-25480: [patch]Add field PauseReason on QueueMemberStatus |
ASTERISK-25481: res_pjsip listens on undefined UDP port, even with no transports configured |
ASTERISK-25482: Faxes are randomly not sent using T.38 when faxing over a local channel |
ASTERISK-25483: [patch] Built-in sounds are send at 40,20,40,20ms intervals for iLBC |
ASTERISK-25484: [patch] autoframing=yes has no effect |
ASTERISK-25485: res_pjsip_outbound_registration: registration stops due to 400 response |
ASTERISK-25486: res_pjsip: Fix deadlock when validating URIs |
ASTERISK-25487: chan_ooh323: Error Decoding H245 Message |
ASTERISK-25488: Mute function issue with Asterisk ARI: Channel not in stasis. |
ASTERISK-25489: Crash while calling function iax2_frame_free |
ASTERISK-25490: [patch]SDP crypto tag is validated incorrectly |
ASTERISK-25491: extenpatternmatchnew=yes breaks with hint and pattern match extension in the same context |
ASTERISK-25492: ARI: Path parameters are case sensitive |
ASTERISK-25493: voicemail email limited to 80 characters |
ASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and missing paren issues |
ASTERISK-25495: [patch] Prevent old-update packages on repository Debian systems |
ASTERISK-25496: Random lockups using ARI |
ASTERISK-25497: Calling fflush on stdout blocking |
ASTERISK-25498: Asterisk crashes when negotiating g729 without that module installed |
ASTERISK-25499: G722 Codec and Inband DTMF Support |
ASTERISK-25500: voicemail update problem in PG when change password |
ASTERISK-25501: Dial processing fail if there is comment like ;--== |
ASTERISK-25502: res_pjsip_pubsub: Subscription not terminated on NOTIFY error responses |
ASTERISK-25503: Compilation failure in Fedora 23 - 'collect2: error: ld returned 1 exit status' |
ASTERISK-25504: Asterisk with pjsip driver crashes codec related? |
ASTERISK-25505: res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created |
ASTERISK-25506: [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. |
ASTERISK-25507: Dual channel redirect causes Volume audiohook to fail to work |
ASTERISK-25508: Defaults cdr.conf.sample |
ASTERISK-25509: Testsuite: tests/rest_api/channels/redirect/nominal Crash |
ASTERISK-25510: [patch]Log to syslog failing |
ASTERISK-25511: Missing Documentation for AMI Events QueueParams and QueueMember |
ASTERISK-25512: Crash every day three times on vmWare |
ASTERISK-25513: Crash: malloc failed with high load of subscriptions. |
ASTERISK-25514: Can'nt compile |
ASTERISK-25515: Testsuite: Handle when AMIFactory does not support reconnections |
ASTERISK-25516: Too many objects of the specified type (PJ_ETOOMANY). Failed to send Request msg OPTIONS. |
ASTERISK-25517: PCMA codec negotiated.. Asterisk unexpectedly uses G722 for first second or so of call before sending PCMA |
ASTERISK-25518: taskprocessor: Add high water mark |
ASTERISK-25519: Confbridge muted user gets unmuted while audio is playback on same conference room |
ASTERISK-25520: ARI DELETE /bridges/{bridgeId} when recording doesn't delete |
ASTERISK-25521: Official web page to read general information about asterisk in spanish? |
ASTERISK-25522: ARI: Crash when creating channel via ARI originate with requesting channel |
ASTERISK-25523: res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. |
ASTERISK-25524: module reload res_calendar.so does not reload everything in calendar.conf |
ASTERISK-25525: A call EndWhile can cause the dial plan to jump to another context. |
ASTERISK-25526: In case of incorrect request in a database, after several attempts, service the asterisk is disconnected |
ASTERISK-25527: Quirky xmldoc description wrapping |
ASTERISK-25528: DNS: System resolver issues with TTL parse |
ASTERISK-25529: (Adaptive)CDR with MySQL Storing LinkedID in Uniqueid column |
ASTERISK-25530: Asterisk 13.6 not presenting ami hangup event for queue calls |
ASTERISK-25531: [patch] add debug detailing the location of a file search for Playback or Background |
ASTERISK-25532: Asterisk crash on certain extension |
ASTERISK-25533: [patch] buffer for ast_format_cap_get_names only 64 bytes |
ASTERISK-25534: Core dump on RTCP report/DNS lookup |
ASTERISK-25535: [patch] format creation on module load instead of cache |
ASTERISK-25536: Testsuite: Sporadic failures in Attended transfer tests using 3PCC |
ASTERISK-25537: [patch] format-attribute module: RFC or internal defaults? |
ASTERISK-25538: [patch]Missing PID in syslog logger messages |
ASTERISK-25539: 488 Not acceptable here sent after T.38 re-invite accepted |
ASTERISK-25540: Core dump related to RTCP report |
ASTERISK-25541: add socket activation support |
ASTERISK-25542: log to journald directly |
ASTERISK-25543: replace openssl with GnuTLS |
ASTERISK-25544: remove upstart from contrib |
ASTERISK-25545: [patch] translation module gets cached not joint format |
ASTERISK-25546: threadpool: Race condition between idle timeout and activation |
ASTERISK-25547: Asterisk crashed when user came in confbridge |
ASTERISK-25548: stasis: Improve message type "Use of before init/after destruction" error |
ASTERISK-25549: Confbridge: Add participant timeout option |
ASTERISK-25550: Codecs negotiated incorrectly |
ASTERISK-25551: [patch]Ability to add channel to an existing bridge by specifying an existing channel prefix |
ASTERISK-25552: hashtab: Improve NULL tolerance |
ASTERISK-25553: endpoints created via pjsip_wizard does not have registrations |
ASTERISK-25554: confusing module loading errors on startup |
ASTERISK-25555: Failing Testsuite Test: 'rls/lists/nominal/presence/full_state' |
ASTERISK-25556: Testsuite: Test Failure Detected for 'rls/lists/nominal/presence/full_state' |
ASTERISK-25557: Testsuite: Add Debug Message for Incorrect Module 'typename' Values Specified in 'test-config.yaml' |
ASTERISK-25558: [patch]chan_sip option 'notifyringing' doc fix and addition of 'notifyringingprio' |
ASTERISK-25559: Asterisk12 with webRTC , can ring but no audio and video |
ASTERISK-25560: speex module fails to compile |
ASTERISK-25561: app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked! |
ASTERISK-25562: testsuite: Ability to per-test disable logging |
ASTERISK-25563: Channel Hangup after DTMF during Playback |
ASTERISK-25564: CHAOS: Assertion in ast_ari_callback |
ASTERISK-25565: DNS: System resolver only returns 1 record per result |
ASTERISK-25566: Double log entries |
ASTERISK-25567: Log IP Addresses for automatic firewalling (e.g. fail2ban) |
ASTERISK-25568: [patch]180 Ringing not sent after 183 Session Progress |
ASTERISK-25569: app_meetme: Audio quality issues |
ASTERISK-25570: DNS: Implement negative connection cache |
ASTERISK-25571: PJSIP: Add StatsD stats for some common PJSIP objects |
ASTERISK-25572: Endpoints: Add StatsD stats for Asterisk endpoints |
ASTERISK-25573: [patch] H.264 format attribute module: resets whole SDP |
ASTERISK-25574: OPening TCP Port 5060 |
ASTERISK-25575: res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart |
ASTERISK-25576: Crash while cancelling thread in chan_skinny |
ASTERISK-25577: Crash while walking thread list to unregister a thread on shutdown |
ASTERISK-25578: [patch] SIP/SDP: No rtpmap for static RTP payload IDs |
ASTERISK-25579: Outgoing Packetization Time (Speex, AMR, Opus, …) |
ASTERISK-25580: WSS will not work with asterisk 11 |
ASTERISK-25581: [patch]Add value reason a pause on CLI |
ASTERISK-25582: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38 |
ASTERISK-25583: [patch] format-attribute module: RFC 7587 (Opus Codec) |
ASTERISK-25584: [patch] format-attribute module: VP8 missing |
ASTERISK-25585: [patch]rasterisk never hits most of main(), but it's assumed to |
ASTERISK-25586: uuid_generate_random detection failure |
ASTERISK-25587: Libedit2 colored prompt is broken beyond repair |
ASTERISK-25588: Problem exchanging device states with PJSIP |
ASTERISK-25589: Improve documentation on transport configuration |
ASTERISK-25590: CLI Usage info for 'pjsip send notify' references incorrect config |
ASTERISK-25591: [patch] Complete List of Header Files (#include): iwyu |
ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at end of a struct |
ASTERISK-25593: fastagi: record file closed after sending result |
ASTERISK-25594: Queue strategy linear |
ASTERISK-25595: Unescaped : in messge sent to statsd |
ASTERISK-25596: CDR engine dispatching 2 cdrs - one for PartyA and another combined PartyA-PartyB |
ASTERISK-25597: Remote console freeze after 'core stop gracefully' and then further command attempts |
ASTERISK-25598: res_pjsip: Contact status messages are printing a hash instead of the uri |
ASTERISK-25599: [patch] SLIN Resampling Codec only 80 msec |
ASTERISK-25600: bridging: Inconsistency in BRIDGEPEER |
ASTERISK-25601: json: Audit reference usage and thread safety |
ASTERISK-25602: chan_sip deadlocks after INVITE processing while calling sip_report_security_event |
ASTERISK-25603: [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash |
ASTERISK-25604: Dial application 'r' argument does not initiate ringing on a Motif channel |
ASTERISK-25605: outboundproxy= works globally but not for individual SIP trunks |
ASTERISK-25606: Core dump when using transports in sorcery |
ASTERISK-25608: res_pjsip/contacts/statsd: Lifecycle events aren't consistent |
ASTERISK-25609: [patch]Asterisk may crash when calling ast_channel_get_t38_state(c) |
ASTERISK-25610: Asterisk crash during "sip reload" |
ASTERISK-25611: core: threadpool thread_timeout_thrash unit test sporadically failing |
ASTERISK-25612: Configuration parser handles unsigned integers as signed integers |
ASTERISK-25613: Documentation: ConnectedLineNum / Name same value as CallerID if call started from ORIGINATE |
ASTERISK-25614: DTLS negotiation delays |
ASTERISK-25615: res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports |
ASTERISK-25616: Warning with a Codec Module which supports PLC with FEC |
ASTERISK-25617: Asterisk 11 segfaults in pj_stun_session_on_rx_pkt |
ASTERISK-25618: res_pjsip: Check for readability of TLS files at startup |
ASTERISK-25619: res_chan_stats not sending the correct information to StatsD |
ASTERISK-25620: Call pickup during a Multi-party Dial results in a channel hanging up later than it should and a duplicate CDR entry. |
ASTERISK-25621: res_pjsip: outbound_proxy arbitrarily and occasionally set to 'asterisk' during reload |
ASTERISK-25622: WARNING for "JABBER: socket read error" should be more specific |
ASTERISK-25623: "><img src=1 onerror=alert(1)> |
ASTERISK-25624: AMI Event OriginateResponse bug |
ASTERISK-25625: res_sorcery_memory_cache: Add full backend caching |
ASTERISK-25626: Integrating Outbound Routes with context |
ASTERISK-25627: Easily Preventable Compile Warning |
ASTERISK-25628: res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging |
ASTERISK-25629: [patch] Native Packet-Loss Concealment (PLC) |
ASTERISK-25630: ari: Creating bridge with id of one that exists outside of ARI doesn't cause error |
ASTERISK-25631: segfault at rtp_engine.c:1525 |
ASTERISK-25632: res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed |
ASTERISK-25633: chan_sip: Codec preference misorder |
ASTERISK-25634: How are updates installed in AsteriskNOW |
ASTERISK-25635: run_agi() while() loop loops indefinitely because of fgets() returns EAGAIN |
ASTERISK-25636: delete user lakatmester |
ASTERISK-25637: Multi homed server using wrong IP |
ASTERISK-25638: pjsip: Deadlock between monitor thread and worker threads |
ASTERISK-25639: app_amd: system maxwords discrepency |
ASTERISK-25640: pbx: Deadlock on features reload and state change hint. |
ASTERISK-25641: bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel |
ASTERISK-25642: res_rtp_asterisk: SRTCP broken with DTLS - bad video is one of the consequences |
ASTERISK-25643: Internal Chan / PJSIP Cause code related Crash |
ASTERISK-25644: No event in queue log when member leaves the queue |
ASTERISK-25645: res_rtp_asterisk: Lock inversion |
ASTERISK-25646: CLI output after "core stop gracefully" on a remote console is confusing and inconsistent with root console behavior |
ASTERISK-25647: bug of cel_radius.c: wrong point of ADD_VENDOR_CODE |
ASTERISK-25648: chan_sip returns forbidden 403, if the incoming number was determined as the present. |
ASTERISK-25649: Transfer application does not work with Local channels - documentation misleading |
ASTERISK-25650: Crash before unload res_pjsip_exten_state.so and load again res_pjsip_exten_state.so |
ASTERISK-25651: Error loading several modules in Fedora 23 with 'undefined symbol' - is modules.conf missing dependencies? |
ASTERISK-25652: func_curl: Add the ability to CURL files down to a specified location |
ASTERISK-25653: Deadlock - PJ_ENOMEM errors & high Recv-Q counts when using PJSIP TLS extensions |
ASTERISK-25654: Playback: Add the ability to play remote URIs |
ASTERISK-25655: core taskprocessors sorcery-control memory not cleared |
ASTERISK-25656: 183 is missing codec mappings with Chan SIP and Asterisk 13 when transcoding required - REINVITE occuring when it shouldn't, also missing codec mappings. |
ASTERISK-25657: pbx: Split up logical parts of the PBX core into separate things |
ASTERISK-25658: Random segmentation fault for asterisk webrtc |
ASTERISK-25659: res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance |
ASTERISK-25660: Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. |
ASTERISK-25661: No clean way to stop Playback in Asterisk 13 and PLAYBACKSTATUS regression |
ASTERISK-25662: Malformed AGI 520 Usage response |
ASTERISK-25663: app_queue: Segfault in queue during playback of periodic announcement - filename address out of bounds |
ASTERISK-25664: ast_format_cap_append_by_type leaks a reference |
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events |
ASTERISK-25666: chan_sip: Path header is ignored |
ASTERISK-25667: Subscription request from endpoint XXX rejected. Expiration of 0 is invalid |
ASTERISK-25668: res_pjsip: Deadlock in distributor |
ASTERISK-25669: [patch]CURL incorrect trim for non ASCII characters |
ASTERISK-25670: Add regcontext to PJSIP |
ASTERISK-25671: Asterisk often gets a SIGSEGV, Segmentation fault |
ASTERISK-25672: 13.6 PJSIP Crash on Startup |
ASTERISK-25673: res_crypto leaks CLI entries |
ASTERISK-25674: Mixmonitor stop recording after atxfer |
ASTERISK-25675: Endpoint not listed as Unreachable |
ASTERISK-25676: chan_sip: Codecs on RTP instance are all offered, not combined |
ASTERISK-25677: pbx_dundi: leaks during failed load. |
ASTERISK-25678: app_confbridge: Add list concise command |
ASTERISK-25679: res_calendar leaks scheduler. |
ASTERISK-25680: manager: manager_channelvars is not cleaned at shutdown |
ASTERISK-25681: devicestate: Engine thread is not shut down |
ASTERISK-25682: Unable to Make with imap support |
ASTERISK-25683: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG |
ASTERISK-25684: codec: Translation of slin16 results in noise |
ASTERISK-25685: infrastructure: Run alembic in Jenkins build script |
ASTERISK-25686: PJSIP: qualify_timeout is a double, database schema is an integer |
ASTERISK-25687: res_musiconhold: Concurrent invocations of 'moh reload' cause a crash |
ASTERISK-25688: configure: No check for PJSIP pj_timer_entry_running |
ASTERISK-25689: pjsip show contacts not working in Asterisk 13.7rc2 |
ASTERISK-25690: Hanging up when executing connected line sub does not cause hangup |
ASTERISK-25691: Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. |
ASTERISK-25692: app_queue: shared_lastcall does not work when using realtime queues and Local channels |
ASTERISK-25693: cdr_pgsql: Refactoring |
ASTERISK-25694: cel_pgsql: Refactoring |
ASTERISK-25695: safe_asterisk -c breaks color in asterisk -r because of missing TERM for /dev/tty. |
ASTERISK-25696: bridge_basic: don't cache xferfailsound during a transfer |
ASTERISK-25697: bridge_basic: don't play an attended transfer fail sound after target hangs up |
ASTERISK-25698: Asterisk stopped |
ASTERISK-25699: Segfault in check_cached_response |
ASTERISK-25700: main/config: Clean config maps on shutdown. |
ASTERISK-25701: core: Endless loop in "core show taskprocessors" |
ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 |
ASTERISK-25703: chan_sip: externrefresh (sip.conf) isn't working |
ASTERISK-25704: pjsip show settings shows debug=no no matter value in debug setting in pjsip.conf |
ASTERISK-25705: PJSIP debug settings are inconsistent between console and file output |
ASTERISK-25706: pbx: Abort asterisk on features reload (handle_hint_change) |
ASTERISK-25707: Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions |
ASTERISK-25708: chan_iax2: No ActionID field in iaxpeers Asterisk Manager command |
ASTERISK-25709: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel |
ASTERISK-25710: core: Crash when publishing message that variable has been set |
ASTERISK-25711: after answering tranfering the call iam disconecting the call ..but still my status is showing as queue |
ASTERISK-25712: Second call to already-on-call phone and Asterisk sends "Ready" |
ASTERISK-25713: Queue produces large number of undesired cdr records |
ASTERISK-25714: ASAN:heap-buffer-overflow in logger.c |
ASTERISK-25715: [patch] ASAN:global-buffer-overflow pjsip |
ASTERISK-25716: Documentation: Document explanations and examples for possible values of DIALSTATUS |
ASTERISK-25717: ASAN in most installed libsrtp |
ASTERISK-25718: file: Use after free during shutdown |
ASTERISK-25719: Direct media failure and strange logger output - similar failures with chan_sip or res_pjsip |
ASTERISK-25720: core: Make channel variable allocation easier to read |
ASTERISK-25721: [patch] res_phoneprov: memory leak and heap-use-after-free |
ASTERISK-25722: ASAN & testsute: stack-buffer-overflow in sip_sipredirect |
ASTERISK-25723: crash on dial with option p or P (privacy mode) |
ASTERISK-25724: Many memory leaks and few asan bugs |
ASTERISK-25725: core: Incorrect XML documentation may result in weird behavior |
ASTERISK-25726: Asterisk compilation fails: 'redhat-hardened-ld: no such file or directory' - Asterisk on Fedora 23 appears to require redhat-rpm-config |
ASTERISK-25727: RPM build requires OPTIONAL_API cflag due to PJSIP requirement |
ASTERISK-25728: Better tracking of calls within Asterisk Logs needed (UNIQUEID) |
ASTERISK-25729: [patch] Extension to device state translations are missing some extension states |
ASTERISK-25730: build: make uninstall after make distclean tries to remove root |
ASTERISK-25731: When a queue agent transfers a queue call, wrapuptime is not respected |
ASTERISK-25732: [patch] persist queue member pause reason through restart |
ASTERISK-25733: Called with SDP without ice-ufrag and ice-pwd |
ASTERISK-25734: Mixmonitor - gaps in recordings |
ASTERISK-25735: [patch] res_xmpp: Does not connect in component mode |
ASTERISK-25736: pbx core: Deadlock during a reload |
ASTERISK-25737: res_pjsip_outbound_registration: line option not in Alembic |
ASTERISK-25738: res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action |
ASTERISK-25739: IAX2 - No native bridge on peers when encryption=yes |
ASTERISK-25740: Asterisk aborts after receiving a MySQL syntax error for adaptive_odbc CDR logging |
ASTERISK-25741: res_pjsip: "Contact" contains UUID for user portion |
ASTERISK-25742: Secondary IFP Packets can result in accessing uninitialized pointers and a crash |
ASTERISK-25743: Registered Peers Goes Unregistered |
ASTERISK-25744: res_pjsip: Segfaults in ssl3_write_bytes, pj_ssl_sock_send, tls_send_msg |
ASTERISK-25745: Crash in append_history_va |
ASTERISK-25746: func_odbc: Requires an additional ARG when executing via CLI and Dialplan |
ASTERISK-25747: Crash on "restart when convenient" |
ASTERISK-25748: No audio h323 Asterisk with Ericsson MD110 |
ASTERISK-25749: StatsD dialplan application not existes |
ASTERISK-25750: features: Crash occurs when executing a "features reload" |
ASTERISK-25751: res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock |
ASTERISK-25752: Register 2 ext cisco 9951 into 2 separate asterisk server |
ASTERISK-25753: No remote Party ID in PJSIP header invite |
ASTERISK-25754: install_prereq doesn't work in Ubuntu 15.10 |
ASTERISK-25756: INVITEs are sent to old peer ip address if usereqphone is set true |
ASTERISK-25757: When the Agents press hold in Softphone (X-litle) automatically the external call are hangup and the System put the Agents logoff. |
ASTERISK-25758: The Callerid don´t appears on X-litle and many Agents after finished is time work cannot logoff |
ASTERISK-25760: SayUnixTime aborts when it tries to give the time in French |
ASTERISK-25761: USAN: Potential runtime errors causing undefined behavior |
ASTERISK-25762: TSAN: Data race json unref |
ASTERISK-25763: TSAN: Data race in json free |
ASTERISK-25764: TSAN: low potencial data race in sig_flags |
ASTERISK-25765: TSAN: data races and lock-order-inversions (potential deadlocks) |
ASTERISK-25766: [patch] USAN can be used together with other sanitizers |
ASTERISK-25767: [patch] Add check to configure for sanitizes |
ASTERISK-25768: astobj2.c:124 INTERNAL_OBJ: bad magic number [..] for object [..] |
ASTERISK-25769: Asterisk 13 is unable to convert audio files to sln48 format |
ASTERISK-25770: Check for OpenSSL defines before trying to use them. |
ASTERISK-25771: ARI:Crash - Attended transfers of channels into Stasis application. |
ASTERISK-25772: res_pjsip: Unexpected two BYE when answered |
ASTERISK-25773: ast flags macros broke atomic thread safe |
ASTERISK-25774: Data race on deleting threads |
ASTERISK-25775: stasis: Race condition with lock destruction in JSON usage |
ASTERISK-25776: lock-order-inversion (potential deadlock) when loading app_queue |
ASTERISK-25777: data race in threadpool |
ASTERISK-25778: lock-order-inversion (potential deadlock) in res_pjsip |
ASTERISK-25779: pjproject: Data race in pj_time |
ASTERISK-25780: stasis: Potential deadlock |
ASTERISK-25781: res_stun_monitor: Potential data race when accessing data |
ASTERISK-25782: data race on logger |
ASTERISK-25783: data race in ast_begin_shutdown |
ASTERISK-25784: lock-order-inversion (potential deadlock) on dialplan reload |
ASTERISK-25785: PJSIP Segmentation fault. |
ASTERISK-25786: How to program with FreeVoip an incoming call on FXO rigning a FXS device with callerID info |
ASTERISK-25787: chan_sip: Race condition when executing "sip show peers" |
ASTERISK-25788: chan_sip: Race condition when executing "sip reload" |
ASTERISK-25789: cdr_adaptive_odbc: Not storing custom fields in all cases |
ASTERISK-25790: Unable to set X-P-Asserted-Identity with PJSIP_HEADER - the header doesn't make it into the SIP packet. |
ASTERISK-25791: res_pjsip_caller_id: Lack of support for Anonymous <anonymous@anonymous.invalid> |
ASTERISK-25792: chan_sip: qualifygap bounds checking |
ASTERISK-25793: chan_sip: Incoming SIP packets ignored |
ASTERISK-25794: Chan_sip allocates RTP ports even for rejected calls |
ASTERISK-25795: Inbound fax is not working for L3 numbers when sent through Ring central account |
ASTERISK-25796: res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES |
ASTERISK-25797: app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension |
ASTERISK-25798: Agent CLI commands no longer exist on 11.21.2 |
ASTERISK-25799: can't place calls to throught sip trunk to cisco |
ASTERISK-25800: [patch] Calculate talktime when is first call answered |
ASTERISK-25801: app_mixmonitor: Does not continue after attended transfer |
ASTERISK-25802: Segfault in rtp_engine.c |
ASTERISK-25803: [patch] chan_sip: Optionally supply fromuser/fromdomain in SIP dial string |
ASTERISK-25804: asterisk crash related to pjsip endpoint reginstration |
ASTERISK-25805: Wiki documentation: Add a basic DUNDi how-to |
ASTERISK-25806: Deep ACD queueing with app_queue.c causes denial of service on systems with a high core count by excessive thread mutex locks |
ASTERISK-25807: Asterisk & WebRTC with DTLS-SRTP |
ASTERISK-25808: Failed unpause update of realtime queue member |
ASTERISK-25809: testsuite: tests/bridge/atxfer_fail_blonde fails or can give a false pass |
ASTERISK-25810: say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. |
ASTERISK-25811: Unable to delete object from sorcery cache |
ASTERISK-25812: res_pjsip_t38: Channel cannot do direct media after T.38 |
ASTERISK-25813: res_config_mysql dbsock parameter |
ASTERISK-25814: Segfault at f ip in res_pjsip_refer.so |
ASTERISK-25815: PJSIP RFC3323 |
ASTERISK-25816: French conf-adminmenu, conf-usermenu prompts differ in content from the English files |
ASTERISK-25817: chan_sip: Keep alive messages contain trailing null byte |
ASTERISK-25818: Blank extensions.conf, refuses to perform "dialplan save" |
ASTERISK-25819: AMI hangup "cause" value ignored or overridden when channel is hungup during process of origination |
ASTERISK-25820: Segmentation fault in ast_channel_dialed_causes_add |
ASTERISK-25821: Segmentation fault in raise() |
ASTERISK-25822: Segmentation fault in _int_malloc() |
ASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. |
ASTERISK-25824: How release a call if Receive a specific reason (e.g. Q.850;cause=7) in 183 Session Progress message |
ASTERISK-25825: Crashes during shutdown when running CLI commands |
ASTERISK-25826: PJSIP / Sorcery slow load from realtime |
ASTERISK-25827: crash asterisk with dialplan add extension |
ASTERISK-25828: Compile failure with older pjproject versions |
ASTERISK-25829: res_pjsip: PJSIP does not accept spaces when separating multiple AORs |
ASTERISK-25830: Revision 2451d4e breaks NAT |
ASTERISK-25831: [patch] CEL MongoDB Backend |
ASTERISK-25832: chan_sip: "Unknown peer" registration error does not retry |
ASTERISK-25833: Asterisk 13.7.2 crashes with error about libmysqlclient.so.18.0.0 |
ASTERISK-25834: [patch] cdr_adaptive_odbc not logger when load before than database |
ASTERISK-25835: Authentication using 'Username' field from Digest |
ASTERISK-25836: Realtime MoH not working |
ASTERISK-25837: file: Blocking when using FIFO |
ASTERISK-25838: elastrix |
ASTERISK-25839: "Expected to acknowledge ticks" problem |
ASTERISK-25840: Asterisk 13.7.0 unable to send INVITEs to jsSIP (WebRTC) peer connected over WSS |
ASTERISK-25841: pbx_spool fails |
ASTERISK-25842: Move from linked lists to another type |
ASTERISK-25843: chan_sip: Registration passwords can not contain @ |
ASTERISK-25844: app_queue: Ghost channels in "core show channels" output |
ASTERISK-25845: res_pjsip_sdp_rtp: Wrong audio codec used when video enabled |
ASTERISK-25846: Gracefully deal with Absent Stasis Apps |
ASTERISK-25847: SLA causing segfaults |
ASTERISK-25848: app_queue: Wrong channel in CONNECT and COMPLETECALLER events when call pickup feature code is used |
ASTERISK-25849: chan_pjsip: transfers with direct media sometimes drops audio |
ASTERISK-25850: Channel related AMI messages are arriving after the CHAN_END and LINKEDID_END CEL messages. |
ASTERISK-25851: Bug in chan_sip - Forbidden 403 |
ASTERISK-25852: chan_iax2: Exceptionally long voice queue length with trunking |
ASTERISK-25853: segfault in libpjnath.so.2 |
ASTERISK-25854: No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk |
ASTERISK-25855: No application AgentLogin after a while of operation |
ASTERISK-25856: chan_sip: Path header is ignored (re-opening) |
ASTERISK-25857: func_aes: incorrect use of strlen() leads to data corruption |
ASTERISK-25858: [patch]Early media not processed when received ACM with some conditions |
ASTERISK-25859: Asterisk not processing the calls |
ASTERISK-25860: app_mixmonitor: Sound distortion on Playback |
ASTERISK-25861: Subscription timeout earlier than anticipated - Asterisk accepts SUBSCRIBE with Event: presence and Accept: application/dialog-info+xml |
ASTERISK-25862: No support for dynamic payload types in direct media |
ASTERISK-25863: Occasionally Asterisk crashes when a iax2 channels connects |
ASTERISK-25864: chan_sip: Error when trying to handle reINVITE from Asterisk in Chrome (DTLS-SRTP related) |
ASTERISK-25865: Message-Account Missing From PJSIP MWI |
ASTERISK-25866: ChanSpy: allow usage of a long queue to store audio frames, to avoid audio loss |
ASTERISK-25867: [patch] Video delay on app_echo |
ASTERISK-25868: Sorcery "append to category" should allow filters |
ASTERISK-25869: chan_sip: "rejected because extension not found" should be logged as a security event |
ASTERISK-25870: Deadlock while using Asterisk over mobile networks |
ASTERISK-25871: Asterisk deadlock when using confbridge |
ASTERISK-25872: logging to syslog despite my configuring not to |
ASTERISK-25873: res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj |
ASTERISK-25874: app_voicemail: Stack buffer overflow in test_voicemail_notify_endl |
ASTERISK-25875: Testsute can't detect builded version of sipp |
ASTERISK-25876: Wiki Documentation - Configuration/Dialplan/Expressions |
ASTERISK-25877: Wiki Documentation - Configuration/Applications/Bridge Application - Examples! |
ASTERISK-25878: Wiki Documentation - Configuration/Applications/FollowMe - create, provide examples |
ASTERISK-25879: Wiki Documentation - Configuration/Applications/Voicemail - Update, add more detail on config options for voicemail.conf, restructure and cleanup IMAP section |
ASTERISK-25880: Wiki Documentation - Configuration/Applications/Queue - Rewrite/add tutorials and examples, remove AEL guide? |
ASTERISK-25881: pbx: Add support for autohints |
ASTERISK-25882: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) |
ASTERISK-25883: In the func_odbc doesn't work the option "escapecommas" |
ASTERISK-25884: unable to ./configure after ./bootstrap.sh |
ASTERISK-25885: res_pjsip: Race condition between adding contact and automatic expiration |
ASTERISK-25886: Not checking for NULL INTERNAL_OBJ object, Asterisk crashes |
ASTERISK-25887: ARI: POST "/channels/{channelId}/continue" inside Gosub - starts at priority n-1, repeats Stasis call before "continuing" |
ASTERISK-25888: Frequent segfaults in function can_ring_entry() of app_queue.c |
ASTERISK-25889: ARI: Add separate "create" and "dial" operations for channels |
ASTERISK-25890: Asterisk 13.8.0 alembic database update fails |
ASTERISK-25891: res_odbc: Crash using ODBC in mysql with heavy usage |
ASTERISK-25892: I am planning to use VMware or KVM for installation of Asterisk using PRI card, kindly let me know does Asterisk support vmware environment |
ASTERISK-25893: Function vmauthenticate accesses uninitialized memory |
ASTERISK-25894: [patch] webrtc video broken due to missing marker bits in RTP streams |
ASTERISK-25895: ODBC configuration not working with Postgres on Debian 8.4_64 |
ASTERISK-25896: app_voicemail: Option 'pollmailboxes' no longer working |
ASTERISK-25897: RTCP feedback broken for video streams |
ASTERISK-25898: Wiki Documentation: Configuration/Functions - create overview and a few child pages discussing the most commonly used functions |
ASTERISK-25899: IMAP access FATAL error: Out of memory |
ASTERISK-25900: PJSIP Endpoint IP Access Controls |
ASTERISK-25901: Add transport for outbound PUBLISH |
ASTERISK-25902: res_sorcery_memory_cache: Crash on "sorcery memory cache expire" CLI command |
ASTERISK-25903: PJSIP AMI Event ContactStatus: add Useragent and RegExpire |
ASTERISK-25904: PJSIP: add contact.updated event |
ASTERISK-25905: Memory leak during perf testing |
ASTERISK-25906: Delete this please. |
ASTERISK-25907: Wiki documentation: Configuration/Interfaces - fill in intro page with a thorough overview |
ASTERISK-25908: Wiki documentation: Deployment - Guides for Asterisk configuration with NAT and firewalls - scope out |
ASTERISK-25909: Wiki documentation: Deployment/Emergency Calling - Create content |
ASTERISK-25910: pjproject: Via headers are not parsed when "received" contains an IPv6 address |
ASTERISK-25911: chan_iax2: IAX Max Retries - hung IAX channels in Ring state - cannot clear channels until Asterisk restart |
ASTERISK-25912: chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set |
ASTERISK-25913: firends and peers cannot connect |
ASTERISK-25914: PJSIP: failed registration with wrong codec name on allow/disallow |
ASTERISK-25915: asterisk error can't have an outgoing call |
ASTERISK-25916: Configuration file processing aborts if a configuration include target does not exist |
ASTERISK-25917: [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself |
ASTERISK-25918: SIP INFO (Key frame requests) not forwarded on |
ASTERISK-25919: Duplicate DTMF events in ARI |
ASTERISK-25920: Asterisk 13.8.0 segfaults using app.queue when ringinuse set to yes and another call comes in. |
ASTERISK-25921: res_odbc: Crash of system in case of function invocation of ODBC |
ASTERISK-25922: res_pjsip_exten_state: Add configuration support for publishing |
ASTERISK-25923: Problem with download |
ASTERISK-25924: chan_pjsip: Polycom SRTP problem |
ASTERISK-25925: Allow Early Bridges on ARI Dials |
ASTERISK-25926: DAHDI PRI calls echo only when no caller ID is present |
ASTERISK-25927: Removed option "registertrying" is still documented in sip.conf.sample |
ASTERISK-25928: res_pjsip: URI validation done outside of PJSIP thread |
ASTERISK-25929: res_pjsip_registrar: AOR_CONTACT_ADDED events not raised |
ASTERISK-25930: PJSIP: disable multi domain to improve realtime performace |
ASTERISK-25931: PJSIP: add "reg_server" to contacts. |
ASTERISK-25932: Error in Ring Strategy "leastrecent" using Dynamic Agents |
ASTERISK-25933: res_pjsip_pubsub: Asterisk ignores expires header value |
ASTERISK-25934: chan_sip should not require sipregs or updateable sippeers table unless rt |
ASTERISK-25935: channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope |
ASTERISK-25936: res_pjsip_dlg_options MODULEINFO section needs to be fixed |
ASTERISK-25937: Send voicemail to yahoo accoun and local account both |
ASTERISK-25938: res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero. |
ASTERISK-25939: Program terminated with SEGV triggered by PJSIP_BYE_METHOD handler |
ASTERISK-25940: AMI PlayDTMF plays DTMF in wrong thread |
ASTERISK-25941: chan_pjsip: Crash on an immediate SIP final response |
ASTERISK-25942: res_pjsip_caller_id: Transfer results in mixed ConnectedLine information |
ASTERISK-25943: chan_local: Exceptionally long voice queue length queuing to Local/1323@internalexten-0001c0df;1 |
ASTERISK-25944: chan_sip: Multiple peers with same name |
ASTERISK-25945: chan_pjsip: ABORT raised in channel_blob_dtor/ast_json_unref |
ASTERISK-25946: queue_log being created even if app_queue is not loaded or existing |
ASTERISK-25947: Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object. |
ASTERISK-25948: ast_pthread_mutex_lock calling ast_reentrancy_lock with lt=0x0 |
ASTERISK-25949: app_followme: FollowMe transmits DTMF tone to caller |
ASTERISK-25950: [patch]SIP channel does not send PeerStatus events for autocreated peers |
ASTERISK-25951: res_agi: run_agi eats frames it shouldn't |
ASTERISK-25952: Hello ,I'm new AGI script in Asterisk and try to run small script that pass parameter from AGI to agi script and return result to dial plan but give me this error AGI Tx >> 510 Invalid or unknown command AGI Script ivrwl.agi completed, returning 0 |
ASTERISK-25953: Segfault in Queue |
ASTERISK-25954: Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName |
ASTERISK-25955: Make single-connection per dns for func_odbc optional |
ASTERISK-25956: Compilation error in conditionally compiled code in config_options.c |
ASTERISK-25957: Segfault in odbc after commit 9b0a96b947437f58fcc88f154ed5080fde529009 |
ASTERISK-25958: tests/app/mixmonitor: Sporadic failure due to incorrect size |
ASTERISK-25959: http_media_cache/retrieve_cache_control_directives: Sporadic failure |
ASTERISK-25960: The config_hook unit test causes Asterisk to crash if run a second time |
ASTERISK-25961: tests/channels/SIP/sip_tls_call: Sporadic crash when running test |
ASTERISK-25962: Asterisk crashes, potential cause: realtime musiconhold |
ASTERISK-25963: func_odbc requires reconnect checks for stale connections |
ASTERISK-25964: Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight |
ASTERISK-25965: res_pjsip_outbound_publish: Allow multiple clients per configuration |
ASTERISK-25966: Database based hints empty during AMI reload or startup |
ASTERISK-25967: testsuite: Fix premature stopping of manager redirect dual tests |
ASTERISK-25968: pjproject_bundled: Configure and make need to be re-tested |
ASTERISK-25969: tests/channels/SIP/sip_bye_also: Sporadic failures |
ASTERISK-25970: Segfault in pjsip_url_compare |
ASTERISK-25971: WebRTC - set Asterisk IP for SDP manually |
ASTERISK-25972: res_pjsip_exten_state: Use body generator to publish extension state |
ASTERISK-25973: Asterisk crashes when call busy agent is enabled |
ASTERISK-25974: Unused realtime MOH classes not purged on 'moh reload' |
ASTERISK-25975: Asterisk 11.22.0 crashes due to error in app_queue.so |
ASTERISK-25976: configs/basic-pbx/asterisk.conf contains incorrect path separator |
ASTERISK-25977: network-manager uninstalled on ubuntu desktop 14.04 |
ASTERISK-25978: res_pjsip_authenticator_digest: Should not use source port in nonce verification |
ASTERISK-25979: res_pjsip: Weird flood of traffic when authentication fails on TCP |
ASTERISK-25980: [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used |
ASTERISK-25981: ARI: PlaybackStopped shows old channel id after attended transfer |
ASTERISK-25982: [patch]res_fax/t38_gateway: Peer V.21 session is created on wrong channel |
ASTERISK-25983: testsuite: Need an install_prereq script |
ASTERISK-25984: res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it |
ASTERISK-25985: chan_sip:sip_poke_peer does't copy port |
ASTERISK-25986: chan_sip: Wrong request line with IMS and Proxy - Asterisk should be using loose routing, but it isn't? |
ASTERISK-25987: testsuite: Correct PJSIP tag to pjsip on 5 attended transfer tests |
ASTERISK-25988: Asterisk is lacking a systemd unit file |
ASTERISK-25989: apps/confbridge: add regcontext feature |
ASTERISK-25990: PJSIP TLS registration should respect client_uri scheme when generating Contact URI |
ASTERISK-25991: ASAN: double free in res_odbc.c |
ASTERISK-25992: How to get all the call status event in c#.net program |
ASTERISK-25993: pjproject: Allow bundling to not require everything it does |
ASTERISK-25994: [patch]res_pjsip: module load priority |
ASTERISK-25995: dpma-firmware.json on downloads.digium.com inconsistent with filenames |
ASTERISK-25996: Remove "live_dangerously" requirement on DB(read) |
ASTERISK-25997: testsuite: Rest API tests that use autobahn fail with versions >= 0.13.1 |
ASTERISK-25998: file: Crash when using nativeformats |
ASTERISK-25999: res_pjsip_dialog_info_body_generator: Remove subscription requirement |